[Asterisk-Users] DTFM tones almost completly muted.

[Peter Bowyer]

On 22/04/05, steve at daviesfam.org <steve at daviesfam.org> wrote:
> 
> 
> On Fri, 22 Apr 2005, Peter Bowyer wrote:
> 
> > On 22/04/05, Ian Hailey <asterisk at dinplug.com> wrote:
> > > Hello everyone,
> > >
> > > I am trying to receive DTMF commands on asterisk from PSTN calls
> > > terminated at my asterisk box. I have tried to terminate the PSTN calls
> > > with both SIP and IAX using sigate.co.uk and voipuser as the PSTN
> > > terminator. When I listen to tones sent from the PSTN side (e.g.
> > > continuous DTMF tone of about 3 seconds) on the asterisk server (stored
> > > in the voice mail) the tone is more or less completely muted, just the
> > > initial tone start can be heard. I am using the G711 codec. Does anyone
> > > have any idea if these tones are on purpose muted by the service
> > > providers or any other reason why it does not work?
> 
> Most likely the DTMF tones have been detected at the point where the call
> was converted PSTN->SIP/IAX, and forwarded instead as an indication (ie
> via SIP INFO or RFC2833 or whatever.  So you won't hear them in a
> recording of the audio stream.  The remaining blip is just the little bit
> at the start before the gateway recognised the tone.
> 
> You should receive the indication in your SIP or IAX connection and
> Asterisk should see it (but its not audio any more).

No, it doesn't work, period. Somehow Sipgate eats the DTMF. No amount
of messing with the DTMFMode settings in Asterisk helps.

Peter

-- 
Peter Bowyer
Email: peter at bowyer.org
Tel: +44 1296 768003
VoIP: sip:peter at bowyer.org